Table Of Contents

Feature Information for Cisco UBE Protocol-Independent Features and Setup

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Note Table 1 lists only the Cisco IOS software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.

Table 1 Feature Information for CUBE Protocol-Independent Features and Setup Features

Cisco Unified Border Element with Gatekeeper is designed to meet the interconnection needs of Internet telephony service providers (ITSPs) and of enterprises. One set of images provides basic interconnection and a second set provides interconnection through an Open Settlement Protocol (OSP) provider, enabling ITSPs to gain the benefits of the Cisco Unified Border Element with Gatekeeper while making use of the routing, billing, and settlement capabilities offered by OSP-based clearinghouses

The SIP event notification mechanism uses NOTIFY messages to signal when certain telephony events take place. In order to send DTMF signals through NOTIFY messages, the gateway notifies the subscriber when DTMF digits are signaled by the originator. The notification contains a message body with a SIP response status line.

This feature is introduced as part of the DTMF Events Through SIP Signaling feature set.

The iLBC is a standard, high-complexity speech codec suitable for robust voice communication over IP. The iLBC has built-in error correction functionality that helps the codec perform in networks with high-packet loss. This codec is supported on both Session Initiation Protocol (SIP) and H.323.

The following commands were introduced or modified: codec ilbc, codec preference,and rtp payload-type.

This feature provides an option for a Secure RTP (SRTP) call to be connected from H.323 to SIP and from SIP to SIP. Additionally, this feature extends SRTP fallback support from the Cisco IOS voice gateway to the Cisco Unified Border Element.

The Support for Negotiation of an Audio Codec from a List of Codecs on Each Leg of a SIP-to-SIP Call on the Cisco Unified Border Element feature supports negotiation of an audio codec using the Voice Class Codec and Codec Transparent infrastructure on the Cisco UBE.

The following command was introduced or modified: voice-class codec (dial peer).

RTP packets are looped back toward the source when the RTP Media Loopback for SIP Calls feature is configured on a dial peer. SIP RTP media loopback helps in verifying the media path between the device originating the call and the intermediate device.

The SIP SRTP Fallback to Nonsecure RTP feature enables a Cisco IOS Session Initiation Protocol (SIP) gateway to fall back from SRTP to RTP by accepting or sending an RTP/AVP(RTP) profile in response to an RTP/SAVP(SRTP) profile. This feature also allows inbound and outbound SRTP calls with nonsecure SIP signaling schemes (such as SIP URL) and provides the administrator the flexibility to configure TLS, IPsec, or any other security mechanism used in the lower layers for secure signaling of crypto attributes.

The following commands were introduced or modified: srtp(voice), srtpnegotiate,and voice-class sip srtp negotiate

The SIP—INFO Method for DTMF Tone Generation feature uses the Session Initiation Protocol (SIP) INFO method to generate dual-tone multifrequency (DTMF) tones on the telephony call leg. SIP methods, or request message types, request a specific action be taken by another user agent (UA) or proxy server. The SIP INFO message is sent along the signaling path of the call.

Universal Transcoding allows transcoding from any suppoted codec to any other supported codec.

VoIP Call Admission Control Using RSVP

12.1(5)T12.2(11)T

Synchronizes RSVP signaling with H.323 Version 2 signaling to ensure that the bandwidth reservation is established in both directions before a call moves to the alerting phase (ringing). This ensures that the called party phone rings only after the resources for the call have been reserved. Using RSVP-based admission control, VoIP applications can reserve network bandwidth and react appropriately if bandwidth reservation fails.

Call Admission Control (CAC) is a deterministic and informed decision that is made before a voice call is established and is based on whether the required network resources are available to provide suitable QoS for the new call.

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Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental.